[OpenSIPS-Users] SRTP to RTP
Dragomir Haralambiev
goup2010 at gmail.com
Thu Aug 1 08:40:12 EDT 2019
Hi,
I check this. All like OK. Here is SIP flow
1. tpengine_offer
UAC1 SRTP ---INVITE ----> Opensips+rtpengine
*audio 4004 RTP/SAVP 8 0 18 101*
$var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin
ICE=remove";
tpengine_offer("$var(rtpengine_flags)");
Opensips+rtpengine ----- INVITE --> UAC2 RTP
*audio 50190 RTP/AVP 8 0 18 101 *
2. rtpengine_answer when receive 183 (Early Media)
*audio 15612 RTP/AVP 0 8 18 101*
Opensips+rtpengine <---- 183 Early Media ---- UAC2 RTP
$var(rtpengine_flags) = "RTP/SAVP replace-session-connection replace-origin
ICE=force";
rtpengine_answer("$var(rtpengine_flags)");
3. rtpengine_answer when receive 200 OK
*audio 15612 RTP/AVP 0 8 18 101*
Opensips+rtpengine <---- 200 OK---- UAC2 RTP
$var(rtpengine_flags) = "RTP/SAVP replace-session-connection replace-origin
ICE=force";
rtpengine_answer("$var(rtpengine_flags)");
UAC1 <----- 200 OK ----- Opensips+rtpengine
*audio 50208 RTP/SAVP 0 8 18 101*
4. tpengine_offer when receive re-INVITE
UAC1 SRTP ---INVITE ----> Opensips+rtpengine
*audio 4004 RTP/SAVP 0 101*
$var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin
ICE=remove";
tpengine_offer("$var(rtpengine_flags)");
Opensips+rtpengine ----- INVITE --> UAC2 RTP
*audio 50190 RTP/AVP 0 101*
На чт, 1.08.2019 г. в 15:16 ч. David Villasmil <
david.villasmil.work at gmail.com> написа:
> You must check your SDPs, verify all going to srtp is indeed SRTP SDP. And
> all going to UAC is not SRTP
>
> On Thu, 1 Aug 2019 at 11:59, Dragomir Haralambiev <goup2010 at gmail.com>
> wrote:
>
>> Hi,
>>
>> 1. tpengine_offer
>>
>> UAC1 SRTP ---INVITE ----> Opensips+rtpengine
>> $var(rtpengine_flags) = "RTP/AVP replace-session-connection
>> replace-origin ICE=remove";
>> tpengine_offer("$var(rtpengine_flags)");
>> Opensips+rtpengine ----- INVITE --> UAC2 RTP
>>
>>
>> 2. rtpengine_answer when receive 183 (Early Media)
>> Opensips+rtpengine <---- 183 Early Media ---- UAC2 RTP
>> $var(rtpengine_flags) = "RTP/SAVP replace-session-connection
>> replace-origin ICE=force";
>> rtpengine_answer("$var(rtpengine_flags)");
>>
>> 3. rtpengine_answer when receive 200 OK
>> Opensips+rtpengine <---- 200 OK---- UAC2 RTP
>> $var(rtpengine_flags) = "RTP/SAVP replace-session-connection
>> replace-origin ICE=force";
>> rtpengine_answer("$var(rtpengine_flags)");
>>
>> 4. tpengine_offer when receive re-INVITE
>> UAC1 SRTP ---INVITE ----> Opensips+rtpengine
>> $var(rtpengine_flags) = "RTP/AVP replace-session-connection
>> replace-origin ICE=remove";
>> tpengine_offer("$var(rtpengine_flags)");
>> Opensips+rtpengine ----- INVITE --> UAC2 RTP
>>
>>
>> In this case UAC1 SRTP not receive voice.
>>
>> Best regards,
>> Dragomir
>>
>> On Wed, Jul 31, 2019, 16:15 David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>>> Hello,
>>>
>>> You need to do this for every leg of the call. This means:
>>>
>>> Call from SRTP client TO non-SRTP:
>>> Remove the ICE, etc.
>>>
>>> When the REPLY with the 200 SDP comes back FROM the non-SRTP, you need
>>> to ADD ICE, etc.
>>>
>>> Hope that makes sense
>>>
>>> David
>>>
>>> On Wed, 31 Jul 2019 at 14:03, Dragomir Haralambiev <goup2010 at gmail.com>
>>> wrote:
>>>
>>>> Hi,
>>>> When change the answer flag to
>>>>
>>>> $var(rtpengine_flags) = " RTP/SAVP rtcp-mux-offer ICE=force";
>>>> rtpengine_answer("$var(rtpengine_flags)");
>>>>
>>>> Call is connected but UAC1 not send and receive voices.
>>>>
>>>> Regards,
>>>>
>>>> Dragomir
>>>>
>>>> На ср, 31.07.2019 г. в 15:53 ч. Sasmita Panda <spanda at 3clogic.com>
>>>> написа:
>>>>
>>>>> Hi Dragomir,
>>>>>
>>>>> I had mentioned to modify this according to your requirement . If
>>>>> your phone only support RTP/SAVP then change the flag what I have
>>>>> mentioned while answering .
>>>>>
>>>>>
>>>>> *Thanks & Regards*
>>>>> *Sasmita Panda*
>>>>> *Senior Network Testing and Software Engineer*
>>>>> *3CLogic , ph:07827611765*
>>>>>
>>>>>
>>>>> On Wed, Jul 31, 2019 at 6:17 PM Johan De Clercq <Johan at democon.be>
>>>>> wrote:
>>>>>
>>>>>> Use rtp/savp
>>>>>>
>>>>>> On Wed, 31 Jul 2019, 14:40 Dragomir Haralambiev, <goup2010 at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>>> Hi,
>>>>>>>
>>>>>>> Thanks for your replay, but this not working.
>>>>>>>
>>>>>>> UAC1 receive 183 session progress with:
>>>>>>> receive audio 50106 UDP/TLS/RTP/SAVP 0 8 18 101
>>>>>>>
>>>>>>> UAC1 send to Opensips CANCEL.
>>>>>>>
>>>>>>> I make test with MicroSips latest version.
>>>>>>>
>>>>>>> Best regards,
>>>>>>> Dragomir
>>>>>>>
>>>>>>> На ср, 31.07.2019 г. в 15:04 ч. Sasmita Panda <spanda at 3clogic.com>
>>>>>>> написа:
>>>>>>>
>>>>>>>> Hi ,
>>>>>>>>
>>>>>>>> You have to do something like below wherever you are calling
>>>>>>>> rtpengine_offer/rtpengine_answer.
>>>>>>>>
>>>>>>>> $var(rtpengine_flags) = "RTP/AVP replace-session-connection
>>>>>>>> replace-origin ICE=remove";
>>>>>>>> rtpengine_offer("$var(rtpengine_flags)");
>>>>>>>>
>>>>>>>> $var(rtpengine_flags) = "UDP/TLS/RTP/SAVP rtcp-mux-offer ICE=force";
>>>>>>>> rtpengine_answer("$var(rtpengine_flags)");
>>>>>>>>
>>>>>>>> You can modify this according to your requirement .
>>>>>>>>
>>>>>>>>
>>>>>>>> *Thanks & Regards*
>>>>>>>> *Sasmita Panda*
>>>>>>>> *Senior Network Testing and Software Engineer*
>>>>>>>> *3CLogic , ph:07827611765*
>>>>>>>>
>>>>>>>>
>>>>>>>> On Wed, Jul 31, 2019 at 5:16 PM Dragomir Haralambiev <
>>>>>>>> goup2010 at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> Hello,
>>>>>>>>>
>>>>>>>>> I have 2 applications connected to Opensips+rtpengine:
>>>>>>>>> UAC1 -use encryption always. SRTP (RTP/SAVP)
>>>>>>>>> UAC2 - never use encryption . RTP (RTP/AVP)
>>>>>>>>>
>>>>>>>>> How to setup Opensips to make follow call:
>>>>>>>>> UAC1 SRTP -----> Opensips+rtpengine -------> UAC2 RTP
>>>>>>>>>
>>>>>>>>> Thanks,
>>>>>>>>> Dragomir
>>>>>>>>> _______________________________________________
>>>>>>>>> Users mailing list
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>>>>>>>>>
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>>> --
>>> Regards,
>>>
>>> David Villasmil
>>> email: david.villasmil.work at gmail.com
>>> phone: +34669448337
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
> --
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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